Hi Folks, I am trying to create a conference enviorment for the following requirement 1.)accepting calls from n no of b-party in such a manner that the voice from b-party is heard to all the users in the conference and vice versa. 2.)making calls from a-party to n-users in such a manner that the voice from a-party is heard to all the users in the conference and vice versa. i have gone through confsample.c to get an idea.However i am successful in registering my account to my sip server but when i try to do pjsip_conf_add_port in my code i get the following error.... 14:23:29.785 pjsua_media.c Media index 0 selected for call 0 14:23:29.785 pjsua_core.c TX 481 bytes Response msg 100/INVITE/cseq=1 (tdta0x212a5f0) to UDP 172.22.0.15:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.22.0.15:5060 ;received=172.22.0.15;branch=z9hG4bKd9ea.2595dbb.1 Via: SIP/2.0/UDP 172.22.25.245:5086 ;rport=5086;branch=z9hG4bK462402ec-1215-e211-87fa-00137237e9b1 Record-Route: <sip:172.22.0.15;lr;did=32f.ff82e9b> Call-ID: 20f600ec-1215-e211-87fa-00137237e9b1 at localhost.localdomain From: "john.smith" <sip:ekigatest@172.22.0.15 >;tag=2efc00ec-1215-e211-87fa-00137237e9b1 To: <sip:2315 at 172.22.0.15> CSeq: 1 INVITE Content-Length: 0 --end msg-- 14:23:29.785 APP Incoming call from "john.smith" < sip:ekigatest at 172.22.0.15>!! 14:23:29.785 strm0x212fd48 VAD temporarily disabled 14:23:29.786 strm0x212fd48 Encoder stream started 14:23:29.786 strm0x212fd48 Decoder stream started 14:23:29.786 pjsua_media.c Media updates, stream #0: speex (sendrecv) 14:23:29.786 pjlib select() I/O Queue created (0x21426e8) 14:23:29.786 pjsua_media.c Opening sound device PCM at 16000/1/20ms &&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&&& 14:23:29.834 ec0x214eef0 AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=100 ms 14:23:29.835 conference.c Port 1 (sip:ekigatest at 172.22.0.15) transmitting to port 0 (Yamaha DS-1 (YMF724F): YMFPCI (hw:0,0)) 14:23:29.835 conference.c Port 0 (Yamaha DS-1 (YMF724F): YMFPCI (hw:0,0)) transmitting to port 1 (sip:ekigatest at 172.22.0.15) 14:23:29.872 os_core_unix.c Info: possibly re-registering existing thread 14:23:30.491 strm0x212fd48 VAD re-enabled Expression 'ret' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1026 Expression 'AlsaOpen( &alsaApi->baseHostApiRep, params, streamDir, &self->pcm )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1183 Expression 'PaAlsaStreamComponent_Initialize( &self->capture, alsaApi, inParams, StreamDirection_In, NULL != callback )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1410 Expression 'PaAlsaStream_Initialize( stream, alsaHostApi, inputParameters, outputParameters, sampleRate, framesPerBuffer, callback, streamFlags, userData )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1991 Expression 'ret' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1026 Expression 'AlsaOpen( &alsaApi->baseHostApiRep, params, streamDir, &self->pcm )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1183 Expression 'PaAlsaStreamComponent_Initialize( &self->capture, alsaApi, inParams, StreamDirection_In, NULL != callback )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1410 Expression 'PaAlsaStream_Initialize( stream, alsaHostApi, inputParameters, outputParameters, sampleRate, framesPerBuffer, callback, streamFlags, userData )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1991 14:23:32.248 *APP Unable to create conference bridge: Device unavailable [code=469984]* Segmentation fault Heres my code...which is called from main.....Please guide me on this ................................. static void on_call_media_state(pjsua_call_id call_id) { pj_caching_pool cp; pjmedia_endpt *med_endpt; pj_pool_t *pool; pjmedia_conf *conf; pj_status_t status; int port_count; pjsua_call_info ci; pjsua_call_get_info(call_id, &ci); pj_caching_pool_init(&cp, &pj_pool_factory_default_policy, 0); /* * Initialize media endpoint. * This will implicitly initialize PJMEDIA too. */ status = pjmedia_endpt_create(&cp.factory, NULL, 1, &med_endpt); //PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1); /* Create memory pool to allocate memory */ pool = pj_pool_create( &cp.factory, /* pool factory */ "wav", /* pool name. */ 4000, /* init size */ 4000, /* increment size */ NULL /* callback on error */ ); if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { // When media is active, connect call to sound device. pjsua_conf_connect(ci.conf_slot, 0); pjsua_conf_connect(0, ci.conf_slot); } status = pjmedia_conf_create( pool, /* pool to use */ 1,/* number of ports */ 44100, 1, 882, 16, 0, /* options */ &conf /* result */ ); if (status != PJ_SUCCESS) app_perror(THIS_FILE, "Unable to create conference bridge", status); // printf("app_perror(THIS_FILE, Unable to create conference bridge\n"); status = pjmedia_conf_add_port(conf,pool,(pjmedia_port*)ci.conf_slot,NULL,NULL); if (status != PJ_SUCCESS) { app_perror(THIS_FILE, "Unable to add conference port", status); printf("Unable to add conference port add port to bridge\n"); //return 1; } } Thanks & Regards John Smith -------------- next part -------------- An HTML attachment was scrubbed... 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